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Voice Feature Guide 01

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Huawei uses machine translation combined with human proofreading to translate this document to different languages in order to help you better understand the content of this document. Note: Even the most advanced machine translation cannot match the quality of professional translators. Huawei shall not bear any responsibility for translation accuracy and it is recommended that you refer to the English document (a link for which has been provided).
Configuring the VoIP PSTN Service (SIP-based)

Configuring the VoIP PSTN Service (SIP-based)

The SIP-based VoIP technology makes the transport network evolve to the IP network without decreasing the voice quality, provides more value-added functions for users, and saves expense.

Application Context

As shown in Figure 1-172, the Access node functions as an SIP access gateway. In the downstream direction, it provides the access to PSTN users; in the upstream direction, it is connected to the IMS system, working with the IMS core to provide the VoIP service based on SIP.
Figure 1-172 Example network of the SIP voice service

Prerequisite

  • The current system protocol is the SIP protocol. If the current system protocol is not the SIP protocol, change the current system protocol to the SIP protocol with reference to the Adding an SIP Interface.
  • According to the actual network, a route from the Access node to the IMS core network device must be configured to ensure that the Access node and the IMS core network device are reachable from each other.
  • The voice daughter board on the control board works in the normal state.
  • Electronic switch 1 must be in location-0 (indicating that the VoIP service is supported) If the SCUB control board is used. For details of the configuration method, see electro-switch.

Data preparation

Table 1-23 provides the data plan for configuring the VoIP service.
Table 1-23 Data plan for configuring the SIP-based VoIP service

Item

Remarks

SIP interface data

(Must be the same as that on the IMS core network device.)

Parameters related to the media stream and the signaling flow

Media and signaling upstream VLAN

It is used as the upstream VLAN of the VoIP service to be configured. Note that the media stream and the signaling stream can use the same VLAN or different VLANs. The result is determined according to the negotiation with the upstream device.

Signaling upstream port

Upstream port for configuring the SIP signaling.

Media IP address and signaling IP address

These IP addresses must be selected from the media and signaling IP address pools. The media and signaling IP address pools consists of all the IP addresses of the Layer 3 interface of the media and signaling upstream VLAN.

Default IP address of the MG

Next hop address from the Access node to the IMS core network device.

NOTICE:
If the media IP address and the signaling IP address are different and the media and the signaling are transmitted upstream through different gateways, ensure that they correspond to correct gateways. Otherwise, normal calls may not be made.

Parameters of the SIP interface

NOTE:

Parameters listed here are mandatory, which means that the SIP interface fails to be enabled if these parameters are not configured.

SIP interface ID

It is the SIP interface ID used for the VoIP service to be configured, which determines the virtual access gateway (VAG) specified for the user.

Signaling port ID of the SIP interface

The value range is 5000-5999. The protocol defines the port ID as 5060.

IP address of the active IMS core network device to which the SIP interface belongs

When dual homing is not configured, parameters of only one IMS core network device are required. If dual homing is configured, the IP address and the port ID of the standby IMS core network device must be configured.

Port ID of the active IMS core network device to which the SIP interface belongs

Transmission mode of the SIP interface

The transmission mode is selected according to the requirements on the IMS core network device. Generally, UDP is adopted.

Home domain of the SIP interface

It corresponds to parameter home-domain in the MG interface attributes.

Index of the profile used by the SIP interface

It corresponds to parameter Profile-index in the MG interface attributes.

IP address obtaining mode of the proxy server

  • In the IP mode, the IP address and the port ID of the active proxy server must be configured.
  • In the DNS-A or DNS-SRV mode, the domain of the active proxy server must be configured.

Ringing mode of the SIP interface

According to the service requirements, the ringing mode of the SIP interface is determined.

Voice user data

(Must be the same as that on the IMS core network device.)

Slot of the voice service board

-

User configuration data

Phone number

The phone number that the IMS core network device allocates to the user must be configured.

User priority

According to the service requirements, user priority needs to be specified. The user priority includes the following:

  • cat1: government1 (category 1 government users)
  • cat2: government2 (category 2 government users)
  • cat3: common (common users)

User type

According to the service requirements, user type needs to be specified. The user type includes the following:

  • DEL: direct exchange lines (default)
  • ECPBX: earth calling PBX
  • LCPBX: loop calling PBX
  • PayPhone: pay phone

System parameters

The system parameters including the overseas version flag and message waiting indication (MWI) mode need to be configured according to the local standards to ensure that the response of the user terminal complies with the local standards.

Overseas parameters

The attributes such as the upper and lower thresholds of the flash-hooking duration need to be configured according to the local standards to ensure that the response of the user terminal complies with the local standards.

PSTN port attributes

If the PSTN port needs to support the polarity reversal accounting, the PSTN port needs to be configured to support the polarity reversal pulse. Other attributes do not need to be modified if there is no special requirement.

Ringing current attributes

You can adjust the ringing volume by modifying the attributes of the ringing current. Generally, the parameters of the ringing current attributes do not need to be modified. You do not need to modify the parameters of the ringing current attributes according to the local standards only when the default ringing current attributes do not meet the local standards.

Procedure

Translation
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Updated: 2019-02-22

Document ID: EDOC1100067358

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